Hi Mark,
I have an Ubuntu box with the package manager version of Asterisk installed, sitting in a data centre connected to the Internet and it works fine.
I have a couple of SIP hand sets, android handets and I have have a SIP trunk to an upstream provider so I can call the rest of the world, not just my private extensions. I most using it for testing not actually voice communications.
Just editing the files by hand isn't difficult but you need to get your head around some of the terminology and the operating model of Asterisk, that confused me a bit at first.
This is an example of how to create a SIP trunk between two Asterisk boxes, creating the SIP peers on each side and adding the dial plan entries (in extensions.conf): http://null.53bits.co.uk/index.php?page=basic-sip-trunk
These are a few basic debugging commands: http://null.53bits.co.uk/index.php?page=cli-examples
Best of luck and have fun!
James.